Supply of music and the like is flourishing nowadays by means of data distribution by MP3 (MPEG1 audio layer 3), FM (Frequency Modulation) broadcasting, voice multiplexing broadcasting and the like. With these means, a data transmission rate (bit/s) changing proportionally with a frequency bandwidth is lowered and the upper frequency limit is lowered by suppressing the high frequency components of a subject audio signal or the like in order to avoid an occupied broad bandwidth and effectively use radio wave resources. For example, if the upper frequency limit is lowered by suppressing the frequency components at about 15 kHz or higher of an audio signal having the upper limit frequency of 20 kHz, the sampling frequency is only ¾ of the original signal frequency so that the data transmission rate can be lowered advantageously. However, it is obvious that an audio signal with suppressed high frequency components has a sound quality inferior to that of the original signal. From this reason, it has been tried to recover approximate suppressed frequency components by some means. In one approach to recover frequency components, a subject signal is distorted to obtain a distorted signal, the frequency band components to be interpolated into the suppressed band are derived from the distorted signal by using a filter, and the frequency band components are added to the target signal to reproduce a signal approximated to the original signal.
In another approach, voice components containing a pair of a fundamental tone and a harmonic tone are derived from an original audio signal, harmonic components on the high frequency side are estimated from the bandwidth of the original audio signal, and the estimated harmonic components are extrapolated relative to the original audio signal.
With the former approach, however, since the waveform of an audio signal is distorted by using a limiter circuit or the like to create harmonics, these harmonics are not necessarily approximate values essentially contained in the original audio signal.
If the latter approach is applied to an original audio signal whose bandwidth of voices or the like was limited, harmonic components of pure sound components cannot be estimated so that extrapolation is impossible. Similarly, sound components whose harmonic components were removed because of a limited bandwidth cannot be estimated and extrapolation is impossible.
In a relatively good approach, a target signal is frequency analyzed, its frequency spectrum pattern is used for estimating the remaining spectrum pattern of suppressed frequency components, and a signal synthesized from these is added to the target signal. Although this approach is excellent in sound quality improvement, there is a practical problem. Namely, it is necessary for this approach to use a short time Fourier transform process and a short time inverse Fourier transform process which are performed at a high resolution over the broad band of a subject signal, resulting in a large amount of computation required for digital signal processing. This leads to requirements for an excessive calculation amount and an excessive circuit scale of a digital signal processor (DSP), lowering a practical value.
In a recently devised approach which proposes a frequency interpolating device and method, the remaining band components of a signal whose frequency components in a particular band were suppressed are derived by using a band-pass filter or the like, frequency-converted and added to the suppressed band wherein the addition level is properly determined from the spectrum envelope information of the remaining frequency components.
Generally, the short time frequency spectrum pattern of a signal has complicated states and its envelope cannot be said that it changes monotonously and smoothly. Therefore, if the intensities of suppressed band components are estimated only from the envelope information and interpolation is performed in a simple manner, a signal not essentially contained in the original signal may be added or an interpolation signal at an excessive level may be added. In this case, the sound quality is not improved but degraded.
The present invention has been made under the above-described circumstances, and aims at providing a signal interpolating device and method having a high practical value capable of recovering an original signal such as an audio signal of high quality from a signal with a suppressed particular frequency band (e.g., high frequency band) of the original signal, providing a very excellent sound quality in terms of auditory senses, and performing signal processing by relatively small scale digital computation.